>From: "Wilson Pickett" <spamsucks2005@gmail.com>
>Date: Wed, 2 May 2007 15:30:21 +0200
>
>Is there a way to do the following scenario?
>
>1) my asterisk box receives an incoming call from a toll free number
>provider such as nufone, voicepulse, etc.
>2) It then dials a number via SIP and outputs a DTMF sequence.
At this point, I assume, the destination SIP has not been invited? The
purpose of the DTMF is either determine which SIP destination to invite or
to perform some other dial plan functions.
>ok, that part we do every day.
>
>3) After DTMF though, is it possible to get the two SIP channels
>(original SIP caller plus SIP called) hooked together and have my pbx
>no longer in the call at all?
>
>tia
If the above is true, then there shouldn't be a problem if all other
conditions for reinvite are satisfied, because Asterisk will only execute
Dial at this point, and that Dial could follow with reinvite. (I assume that
the original SIP caller is in fact the toll free provider.)
Yuan Liu